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<?xml version="1.0" encoding="iso-8859-1"?> <!-- -*-html-helper-*- -->
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
<html xmlns="http://www.w3.org/1999/xhtml">
<head>
<title>polypaudio: Loadable Modules</title>
<link rel="stylesheet" type="text/css" href="style.css" />
</head>
<body>
<h1>Loadable Modules</h1>
<p>The following loadable modules are provided with the <tt>polypaudio</tt> distribution:</p>
<h2>Device Drivers</h2>
<p>All device driver modules support the following parameters:</p>
<table>
<tr><td><tt>format=</tt></td><td>The sample format (one of <tt>u8</tt>, <tt>s16</tt>, <tt>s16le</tt>, <tt>s16le</tt>, <tt>float32</tt>, <tt>float32be</tt>, <tt>float32le</tt>, <tt>alaw</tt>, <tt>ulaw</tt>) (defaults to <tt>s16</tt>)</td></tr>
<tr><td><tt>rate=</tt></td><td>The sample rate (defaults to 44100)</td></tr>
<tr><td><tt>channels=</tt></td><td>Audio channels (defaults to 2)</td></tr>
<tr><td><tt>sink_name=</tt>, <tt>source_name=</tt></td><td>Name for the sink (resp. source)</td></tr>
<tr><td><tt>channel_map=</tt></td><td>Channel map. A list of
comma-seperated channel names. The currently defined channel names
are: <tt>left</tt>, <tt>right</tt>, <tt>mono</tt>, <tt>center</tt>,
<tt>front-left</tt>, <tt>front-right</tt>, <tt>front-center</tt>,
<tt>rear-center</tt>, <tt>rear-left</tt>, <tt>rear-right</tt>,
<tt>lfe</tt>, <tt>subwoofer</tt>, <tt>front-left-of-center</tt>,
<tt>front-right-of-center</tt>, <tt>side-left</tt>,
<tt>side-right</tt>, <tt>aux0</tt>, <tt>aux1</tt> to <tt>aux15</tt>,
<tt>top-center</tt>, <tt>top-front-left</tt>,
<tt>top-front-right</tt>, <tt>top-front-center</tt>,
<tt>top-rear-left</tt>, <tt>top-rear-right</tt>,
<tt>top-rear-center</tt>, (Default depends on the number of channels
and the driver)</td></tr> </table>
<h3>module-pipe-sink</h3>
<p>Provides a simple test sink that writes the audio data to a FIFO
special file in the file system. The sink name defaults to <tt>pipe_output</tt>.</p>
<p>The following option is supported:</p>
<table>
<tr><td><tt>file=</tt></td><td>The name of the FIFO special file to use. (defaults to: <tt>/tmp/music.output</tt>)</td></tr>
</table>
<h3>module-pipe-source</h3>
<p>Provides a simple test source that reads the audio data from a FIFO
special file in the file system. The source name defaults to <tt>pipe_input</tt>.</p>
<p>The following option is supported:</p>
<table>
<tr><td><tt>file=</tt></td><td>The name of the FIFO special file to use. (defaults to: <tt>/tmp/music.input</tt>)</td></tr>
</table>
<h3>module-null-sink</h3>
<p>Provides a simple null sink. All data written to this sink is silently dropped. This sink is clocked using the system time.</p>
<p>This module doesn't support any special parameters</p>
<a name="module-alsa-sink"/>
<h3>module-alsa-sink</h3>
<p>Provides a playback sink for devices supported by the <a href="http://www.alsa-project.org/">Advanced Linux
Sound Architecture</a> (ALSA). The sink name defaults to <tt>alsa_output</tt>.</p>
<p>In addition to the general device driver options described above this module supports:</p>
<table>
<tr><td><tt>device=</tt></td><td>The ALSA device to use. (defaults to "plughw:0,0")</td></tr>
<tr><td><tt>fragments=</tt></td><td>The desired fragments when opening the device. (defaults to 12)</td></tr>
<tr><td><tt>fragment_size=</tt></td><td>The desired fragment size in bytes when opening the device (defaults to 1024)</td></tr>
</table>
<h3>module-alsa-source</h3>
<p>Provides a recording source for devices supported by the Advanced
Linux Sound Architecture (ALSA). The source name defaults to <tt>alsa_input</tt>.</p>
<p>This module supports <tt>device=</tt>, <tt>fragments=</tt> and <tt>fragment_size=</tt> arguments the same way as <a href="#module-alsa-sink"><tt>module-alsa-sink</tt></a>.</p>
<a name="module-oss"/>
<h3>module-oss</h3>
<p>Provides both a sink and a source for playback, resp. recording on
<a href="http://www.opensound.com">Open Sound System</a> (OSS) compatible devices.</p>
<p>This module supports <tt>device=</tt> (which defaults to <tt>/dev/dsp</tt>), <tt>fragments=</tt> and <tt>fragment_size=</tt> arguments the same way as <a href="#module-alsa-sink"><tt>module-alsa-sink</tt></a>.</p>
<p>In addition this module supports the following options:</p>
<table>
<tr><td><tt>record=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the recording on this device. (defaults: to 1)</td></tr>
<tr><td><tt>playback=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the playback on this device. (defaults: to 1)</td></tr>
</table>
<p>The sink name (resp. source name) defaults to <tt>oss_output</tt> (resp. <tt>oss_input</tt>).</p>
<h3>module-oss-mmap</h3>
<p>Similar to <tt>module-oss</tt> but uses memory mapped
(<tt>mmap()</tt>) access to the input/output buffers of the audio
device. This provides better latency behaviour but is not as
compatible as <tt>module-oss</tt>.</p>
<p>This module accepts exactly the same arguments as <a href="#module-oss"><tt>module-oss</tt></a>.</p>
<h3>module-solaris</h3>
<P>Provides a sink and source for the Solaris audio device.</p>
<p>In addition to the general device driver options described above this module supports:</p>
<table>
<tr><td><tt>record=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the recording on this device. (defaults: to 1)</td></tr>
<tr><td><tt>playback=</tt></td><td>Accepts a binary numerical value for enabling (resp. disabling) the playback on this device. (defaults: to 1)</td></tr>
<tr><td><tt>buffer_size=</tt></td><td>Record buffer size</td></tr>
</table>
<h3>module-waveout</h3>
<P>Provides a sink and source for the Win32 audio device.</p>
<p>This module supports all arguments thet <tt>module-oss</tt> supports except <tt>device=</tt>.</p>
<a name="module-combine"/>
<h3>module-combine</h3>
<p>This combines two or more sinks into one. A new virtual sink is
allocated. All data written to it is forwarded to all connected
sinks. In aequidistant intervals the sample rates of the output sinks
is recalculated: i.e. even when the sinks' crystals deviate (which is
normally the case) output appears synchronously to the human ear. The
resampling required for this may be very CPU intensive.</p>
<table>
<tr><td><tt>sink_name=</tt></td><td>The name for the combined sink. (defaults to <tt>combined</tt>)</td></tr>
<tr><td><tt>master=</tt></td><td>The name of the first sink to link into the combined think. The sample rate/type is taken from this sink.</td></tr>
<tr><td><tt>slaves=</tt></td><td>Name of additional sinks to link into the combined think, seperated by commas.</td></tr>
<tr><td><tt>adjust_time=</tt></td><td>Time in seconds when to readjust the sample rate of all sinks. (defaults to 20)</td></tr>
<tr><td><tt>resample_method=</tt></td><td>Resampling algorithm to
use. See <tt>libsamplerate</tt>'s documentation for more
information. Use one of <tt>sinc-best-quality</tt>,
<tt>sinc-medium-quality</tt>, <tt>sinc-fastest</tt>,
<tt>zero-order-hold</tt>, <tt>linear</tt>. If the default happens to
be to slow on your machine try using <tt>zero-order-hold</tt>. This
will decrease output quality however. (defaults to
<tt>sinc-fastest</tt>)</td></tr> </table>
<h3>module-tunnel-{sink,source}</h3>
<p>Tunnel a remote sink/source to a local "ghost"
sink/source. Requires a running polypaudio daemon on the remote server
with <tt>module-native-protocol-tcp</tt> loaded. It's probably a
better idea to connect to the remote sink/source directly since some
buffer control is lost through this tunneling.</p>
<table>
<tr><td><tt>server=</tt></td><td>The server to connect to</td></tr>
<tr><td><tt>source=</tt></td><td>The source on the remote server. Only available for <tt>module-tunnel-source</tt>.</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink on the remote server. Only available for <tt>module-tunnel-sink</tt>.</td></tr>
<tr><td><tt>cookie=</tt></td><td>The authentication cookie file to use.</td></tr>
</table>
<h3>module-esound-sink</h3>
<p>Create a playback sink using an <a href="http://www.tux.org/~ricdude/apps.html">ESOUND</a> server as backend. Whenever you can, try to omit this
module since it has many disadvantages including bad latency
and even worse latency measurement. </p>
<table>
<tr><td><tt>server=</tt></td><td>The server to connect to</td></tr>
<tr><td><tt>cookie=</tt></td><td>The authentication cookie file to use.</td></tr>
</table>
<h2>Protocols</h2>
<a name="module-cli"/>
<h3>module-cli</h3>
<p>Provides the user with a simple command line interface on the
controlling TTY of the daemon. This module may not be loaded more than
once.</p>
<p>For an explanation of the simple command line language used by this
module see <a href="cli.html"><tt>cli.html</tt></a>.
<p>This module doesn't accept any arguments.</p>
<a name="module-cli-protocol-unix"/>
<a name="module-cli-protocol-tcp"/>
<a name="module-cli-protocol"/>
<h3>module-cli-protocol-{unix,tcp}</h3>
<p>An implemenation of a simple command line based protocol for
controlling the <tt>polypaudio</tt> daemon. If loaded, the user may
connect with tools like <tt>netcat</tt>, <tt>telnet</tt> or
<a href="http://0pointer.de/lennart/projects/bidilink/"><tt>bidilink</tt></a> to the listening sockets and execute commands the
same way as with <tt>module-cli</tt>.</p>
<p><b>Beware!</b> Users are not authenticated when connecting to this
service.</p>
<p>This module exists in two versions: with the suffix <tt>-unix</tt>
the service will listen on an UNIX domain socket in the local file
system. With the suffix <tt>-tcp</tt> it will listen on a network
transparent TCP/IP socket. (Both IPv6 and IPv4 - if available)</p>
<p>This module supports the following options:</p>
<table>
<tr><td><tt>port=</tt></td><td>(only for <tt>-tcp</tt>) The port number to listen on (defaults to 4712)</td></tr>
<tr><td><tt>loopback=</tt></td><td>(only for <tt>-tcp</tt>) Accepts
a numerical binary value. If 1 the socket is bound to the loopback
device, i.e. not publicly accessible. (defaults to 1)</td></tr>
<tr><td><tt>listen=</tt></td><td>(only for <tt>-tcp</tt>) The IP address to listen on. If specified, supersedes the value specified in <tt>loopback=</tt></td></tr>
<tr><td><tt>socket=</tt></td><td>(only for <tt>-unix</tt>) The UNIX socket name (defaults to <tt>/tmp/polypaudio/cli</tt>)</td></tr>
</table>
<h3>module-simple-protocol-{unix,tcp}</h3>
<p>An implementation of a simple protocol which allows playback by using
simple tools like <tt>netcat</tt>. Just connect to the listening
socket of this module and write the audio data to it, or read it from
it for playback, resp. recording.</p>
<p><b>Beware!</b> Users are not authenticated when connecting to this
service.</p>
<p>See <tt>module-cli-protocol-{unix,tcp}</tt> for more information
about the two possible suffixes of this module.</p>
<p>In addition to the options supported by <a href="module-cli-protocol"><tt>module-cli-protocol-*</tt></a>, this module supports:</p>
<table>
<tr><td><tt>rate=</tt>, <tt>format=</tt>, <tt>channels=</tt></td><td>Sample format for streams connecting to this service.</td></tr>
<tr><td><tt>playback=</tt>, <tt>record=</tt></td><td>Enable/disable playback/recording</td></tr>
<tr><td><tt>sink=</tt>, <tt>source=</tt></td><td>Specify the sink/source this service connects to</td></tr>
</table>
<h3>module-esound-protocol-{unix,tcp}</h3>
<p>An implemenation of a protocol compatible with the <a
href="http://www.tux.org/~ricdude/EsounD.html">Enlightened Sound
Daemon</a> (ESOUND, <tt>esd</tt>). When you load this module you may
access the <tt>polypaudio</tt> daemon with tools like <tt>esdcat</tt>,
<tt>esdrec</tt> or even <tt>esdctl</tt>. Many applications, such as
XMMS, include support for this protocol.</p>
<p>See <tt>module-cli-protocol-{unix,tcp}</tt> for more information
about the two possible suffixes of this module.</p>
<p>In addition to the options supported by <a href="module-cli-protocol"><tt>module-cli-protocol-*</tt></a>, this module supports:</p>
<table>
<tr><td><tt>sink=</tt>, <tt>source=</tt></td><td>Specify the sink/source this service connects to</td></tr>
<tr><td><tt>auth-anonymous=</tt></td><td>If set to 1 no authentication is required to connect to the service</td></tr>
<tr><td><tt>cookie=</tt></td><td>Name of the cookie file for authentication purposes</td></tr>
</table>
<p>This implementation misses some features the original ESOUND has: e.g. there is no sample cache yet. However: XMMS works fine.</p>
<h3>module-native-protocol-{unix,tcp}</h3>
<p>The native protocol of <tt>polypaudio</tt>.</p>
<p>See <tt>module-cli-protocol-{unix,tcp}</tt> for more information
about the two possible suffixes of this module.</p>
<p>In addition to the options supported by <a href="module-cli-protocol"><tt>module-cli-protocol-*</tt></a>, this module supports:</p>
<table>
<tr><td><tt>auth-anonymous=</tt></td><td>If set to 1 no authentication is required to connect to the service</td></tr>
<tr><td><tt>auth-group=</tt></td><td>(only for <tt>-unix</tt>): members of the specified unix group may access the server without further auhentication.</td></tr>
<tr><td><tt>cookie=</tt></td><td>Name of the cookie file for authentication purposes</td></tr>
</table>
<h3>module-native-protocol-fd</h3>
<p>This is used internally when auto spawning a new daemon. Don't use it directly.</p>
<h3>module-http-protocol-tcp</h3>
<p>A proof-of-concept HTTP module, which can be used to introspect
the current status of the polypaudio daemon using HTTP. Just load this
module and point your browser to <a
href="http://localhost:4714/">http://localhost:4714/</a>. This module takes the same arguments
as <tt>module-cli-protocol-tcp</tt>.</p>
<h2>X Window System</h2>
<h3>module-x11-bell</h3>
<p>Intercepts X11 bell events and plays a sample from the sample cache on each occurence.</p>
<table>
<tr><td><tt>display=</tt></td><td>X11 display to connect to. If ommited defaults to the value of <tt>$DISPLAY</tt></td></tr>
<tr><td><tt>sample=</tt></td><td>The sample to play. If ommited defaults to <tt>x11-bell</tt>.</td></tr>
<tr><td><tt>sink=</tt></td><td>Name of the sink to play the sample on. If ommited defaults to the default sink.</td></tr>
</table>
<h3>module-x11-publish</h3>
<p>Publishes the access credentials to the Polypaudio server in the
X11 root window. The following properties are used:
<tt>POLYP_SERVER</tt>, <tt>POYLP_SINK</tt>, <tt>POLYP_SOURCE</tt>,
<tt>POLYP_COOKIE</tt>. This is very useful when using SSH or any other
remote login tool for logging into other machines and getting audio
playback to your local speakers. The Polypaudio client libraries make
use of this data automatically. Instead of using this module you may
use the tool <tt>pax11publish</tt> which may be used to access, modify
and import credential data from/to the X11 display.</p>
<table>
<tr><td><tt>display=</tt></td><td>X11 display to connect to. If ommited defaults to the value of <tt>$DISPLAY</tt></td></tr>
<tr><td><tt>sink=</tt></td><td>Name of the default sink. If ommited this property isn't stored in the X11 display.</td></tr>
<tr><td><tt>source=</tt></td><td>Name of the default source. If ommited this property isn't stored in the X11 display.</td></tr>
<tr><td><tt>cookie=</tt></td><td>Name of the cookie file of the
cookie to store in the X11 display. If ommited the cookie of an
already loaded protocol module is used.</td></tr> </table>
<h2>Volume Control</h2>
<h3>module-mmkbd-evdev</h3>
<p>Adjust the volume of a sink when the special multimedia buttons of modern keyboards are pressed.</p>
<table>
<tr><td><tt>device=</tt></td><td>Linux input device ("<tt>evdev</tt>", defaults to <tt>/dev/input/event0</tt>)</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
</table>
<h3>module-lirc</h3>
<p>Adjust the volume of a sink when the volume buttons of an infrared remote control are pressed (through LIRC).</p>
<table>
<tr><td><tt>config=</tt></td><td>The LIRC configuration file</td></tr>
<tr><td><tt>appname=</tt></td><td>The application name to pass to LIRC (defaults to <tt>polypaudio</tt>)</td></tr>
<tr><td><tt>sink=</tt></td><td>The sink to control</td></tr>
</table>
<a name="rtp"/>
<h2>RTP/SDP/SAP Transport</h2>
<p>Polypaudio can stream audio data to an IP multicast group via the
standard protocols <a
href="http://en.wikipedia.org/wiki/Real-time_Transport_Protocol">RTP</a>,
<a
href="http://en.wikipedia.org/wiki/Session_Announcement_Protocol">SAP</a>
and <a
href="http://en.wikipedia.org/wiki/Session_Description_Protocol">SDP</a>
(RFC3550, RFC3551, RFC2327, RFC2327). This can be used for multiple
different purposes: for sharing a single microphone on multiple
computers on the local LAN, for streaming music from a single
controlling PC to multiple PCs with speakers or to implement a simple
"always-on" teleconferencing solution.</p>
<p>The current implementation is designed to be used exlusively in
local area networks, though Internet multicasting is theoretically
supported. Only uncompressed audio is supported, hence you won't be
able to multicast more than a few streams at the same time over a
standard LAN.</p>
<p>Polypaudio implements both a sender and a reciever for RTP
traffic. The sender announces itself via SAP/SDP on the same multicast
group as it sends the RTP data to. The reciever picks up the SAP/SDP
announcements and creates a playback stream for each
session. Alternatively you can use any RTP capable client to
recieve and play back the RTP data (such as <tt>mplayer</tt>).</p>
<h3>module-rtp-send</h3>
<p>This is the sender side of the RTP/SDP/SAP implementation. It reads
audio data from an existing source and forwards it to the network
encapsulated in RTP. In addition it sends SAP packets with an SDP
session description.</p>
<p>In combination with the monitor source of <tt>module-null-sink</tt>
you can use this module to create an RTP sink.</p>
<table>
<tr><td><tt>source=</tt></td><td>The source to read the audio data from. If ommited defaults to the default source.</td></tr>
<tr><td><tt>format=, rate=, channels=</tt></td><td>Sample format to use, defaults to the source's.</td></tr>
<tr><td><tt>destination=</tt></td><td>Destination multicast group for both RTP and SAP packets, defaults to <tt>224.0.0.56</tt></td></tr>
<tr><td><tt>port=</tt></td><td>Destination port number of the RTP
traffic. If ommited defaults to a randomly chosen even port
number. Please keep in mind that the RFC suggests to use only even
port numbers for RTP traffic.</td></tr>
<tr><td><tt>mtu=</tt></td><td>Maximum payload size for RTP packets. If ommited defaults to 1280</td></tr>
<tr><td><tt>loop=</tt></td><td>Takes a boolean value, specifying whether locally generated RTP traffic should be looped back to the local host. Disabled by default.</td></tr>
</table>
<h3>module-rtp-recv</h3>
<p>This is the reciever side of the RTP/SDP/SAP implementation. It
picks up SAP session announcements and creates an RTP playback stream
for each.</p>
<p>In combination with <tt>module-null-sink</tt> you can use this
module to create an RTP source.</p>
<table>
<tr><td><tt>sink=</tt></td><td>The sink to connect to. If ommited defaults to the default sink.</td></tr>
<tr><td><tt>sap_address=</tt></td><td>The multicast group to join for SAP announcements, defaults to <tt>224.0.0.56</tt>.</td></tr>
</table>
<h2>JACK Connectivity</h2>
<p>Polypaudio can be hooked up to a <a
href="http://jackit.sourceforge.net/">JACK Audio Connection Kit</a> server which is a specialized sound server used for professional audio production on Unix/Linux. Both a
Polypaudio sink and a source are available. For each channel a port is
created in the JACK server.</p>
<h3>module-jack-sink</h3>
<p>This module implements a Polypaudio sink that connects to JACK and registers as many output ports as requested.</p>
<table>
<tr><td><tt>sink_name=</tt></td><td>The name for the Polypaudio sink. If ommited defaults to <tt>jack_out</tt>.</td></tr>
<tr><td><tt>server_name=</tt></td><td>The JACK server to connect to. If ommited defaults to the default server.</td></tr>
<tr><td><tt>client_name=</tt></td><td>The client name to tell the JACK server. If ommited defaults to <tt>polypaudio</tt>.</td></tr>
<tr><td><tt>channels=</tt></td><td>Number of channels to register. If ommited defaults to the number of physical playback ports of the JACK server.</td></tr>
<tr><td><tt>connect=</tt></td><td>Takes a boolean value. If enabled (the default) Polypaudio will try to connect its ports to the physicial playback ports of the JACK server</td></tr>
</table>
<h3>module-jack-source</h3>
<p>This module implements a Polypaudio source that connects to JACK
and registers as many input ports as requested. Takes the same
arguments as <tt>module-jack-sink</tt>, except for <tt>sink_name</tt>
which is replaced by <tt>source_name</tt> (with a default of <tt>jack_in</tt>) for obvious reasons.</p>
<h2>Miscellaneous</h2>
<h3>module-sine</h3>
<p>Creates a sink input and generates a sine waveform stream.</p>
<table>
<tr><td><tt>sink=</tt></td><td>The sink to connect to. If ommited defaults to the default sink.</td></tr>
<tr><td><tt>frequency=</tt></td><td>The frequency to generate in Hertz. Defaults to 440.</td></tr>
</table>
<h3>module-esound-compat-spawnfd</h3>
<p>This is a compatibility module for <tt>libesd</tt> based autospawning of polypaudio. Don't use it directly.</p>
<h3>module-esound-compat-spawnpid</h3>
<p>This is a compatibility module for <tt>libesd</tt> based autospawning of polypaudio. Don't use it directly.</p>
<h3>module-match</h3>
<p>Adjust the volume of a playback stream automatically based on its name.</p>
<table>
<tr><td><tt>table=</tt></td><td>The regular expression matching table file to use (defaults to <tt>~/.polypaudio/match.table</tt>)</td></tr>
</table>
<p>The table file should contain a regexp and volume on each line, seperated by spaces. An example:</p>
<pre>
^sample: 32000
</pre>
<p>The volumes of all streams with titles starting with <tt>sample:</tt> are automatically set to 32000. (FYI: All sample cache streams start with <tt>sample:</tt>)</p>
<h3>module-volume-restore</h3>
<p>Adjust the volume of a playback stream automatically based on its name.</p>
<table>
<tr><td><tt>table=</tt></td><td>The table file to use (defaults to <tt>~/.polypaudio/volume.table</tt>)</td></tr>
</table>
<p>In contrast to <tt>module-match</tt> this module needs no explicit
configuration. Instead the volumes are saved and restored in a fully
automatical fashion depending on the client name to identify
streams. The volume for a stream is automatically saved every time it is
changed and than restored when a new stream is created.</p>
<h3>module-detect</h3>
<p>Automatically detect the available sound hardware and load modules for it. Supports OSS, ALSA, Solaris and Win32 output drivers.
<table>
<tr><td><tt>just-one=</tt></td><td>If set to <tt>1</tt> the module will only try to load a single sink/source and than stop.</td></tr>
</table>
<h3>module-zeroconf-publish</h3>
<p>Publish all local sinks/sources using mDNS Zeroconf.</p>
<hr/>
<address class="grey">Lennart Poettering <@PACKAGE_BUGREPORT@>, April 2006</address>
<div class="grey"><i>$Id$</i></div>
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